• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.15 no.5 s.98
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• pp.571-577
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• 2005

The acoustic absorption of multiple layer perforated panel systems is largely reduced at the anti-resonance frequency. In order to improve the acoustic absorption at the anti-resonance frequency, the sound absorbing materials are inserted between perforated panels. By the insertion of absorbing materials, it is found that the multiple layer perforated panel system has better acoustic absorption at the anti-resonance frequency and more broadband frequency. Besides, it is shown that the absorption coefficients from the transfer matrix method agree well with the values measured by the two-microphone impedance tube method for various combinations of perforated panels, airspaces or sound absorbing materials.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.1
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• pp.71-79
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• 2008

A helmholtz resonator has been widely used for the purpose of suppressing the low frequency noises propagated from various heat and fluid machineries. However, the conventional resonator has demerits that the effective absorption bandwidth is narrow and the absorption performance is not so outstanding in the only limited configurations of neck and cavity as well. In order to overcome these problems, in this paper, a resonator with perforated neck is proposed. The absorption performances of the resonator are measured by two-microphone method and estimated by transfer matrix method. The measured values of normal absorption coefficients agree well with the estimated values. By introducing the perforated plates at the neck of a resonator, it is shown that the absorption performance have been significantly improved.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.806-811
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• 1994

Two kinds of static and dynamic state estimation methods are newly discussed for the problem of the measurement disturbance of environmental low-frequency noise in the presence of wind-induced noise. First, the probability characteristics of wind-induced noise are discussed in the form of probability distribution conditioned by wind speed, based on the simultaneous observation of the wind-induced noise and wind speed near a microphone. Next, especially form the viewpoint of simplicity for practical use, two kinds of static and dynamic state estimation methods are discussed. The static estimation method using the information on wind speed is fundamentally supported by the conservation principle of energy sum. The dynamic one is the method by using a recursive digital filter with the parameters successively renewed by the information on wind speed. This can be also simplified by using well-know Kalman filter under the assumption of the Gaussian distribution. The effectiveness of proposed two estimation methods are shown through experiments under a breezy condition in the open filed.

• The Journal of the Acoustical Society of Korea
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• v.36 no.5
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• pp.345-352
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• 2017

We propose a methods which is reducing direction estimation error of sound source in the reverberant and noisy environments. The proposed algorithm divides speech signal into voice and unvoice using VAD. We estimate the direction of source when current frame is voiced. TDOA (Time-Difference of Arrival) between microphone array using the GCC-PHAT (Generalized Cross Correlation with Phase Transform) method will be estimated in that frame. Then, we compare the peak value of cross-correlation of two signals applied to estimated time-delay with other time-delay in time-table in order to improve the accuracy of source location. If the angle of current frame is far different from before and after frame in successive voiced frame, the angle of current frame is replaced with mean value of the estimated angle in before and after frames.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.36-42
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• 2001

This paper propose new sound localization algorithm that detects the sound source bearing in a closed office environment using two microphone array. The proposed Subband CPSP (Cross Power Spectrum Phase) algorithm is a development of previously Down CPSP method using subband approach. It first split the received microphone signals into subbands and then calculates subband CPSP which result in possible source bearings. This type of algorithm, Subband CPSP, can provide more robust and reliable sound localization system because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, a real time simulation was conducted and it was compared with previous CPSP method. From the simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than 5％ average accuracy for sound source detection.

• Journal of KSNVE
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• v.4 no.3
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• pp.295-305
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• 1994

The reflection coefficient of a material at oblique incidence is measured in a free field. The sound pressure distributions are measured at discrete points on two measurement lines and then decomposed into plane wave components by using spatial Fourier transform. The inciedent and reflected plane wave components are obtained from a set of "decomposition equations" of which uses the plane wave propagation theory. Numerical simulations and experiments have been performed to see the effect of finite size of measurement area. To reduce this effect, a window fuction has been performed to see the effects of finite size of mesurement area. To reduce this effect, a window function has been proposed and its effect on the measurement of sound absorbing material property has been studied as well. The reflection coefficient obtained by this method is compared with those obtained from other methods; 2-microphone method in a duct and an expirical equation of which determines the characteristic impedance .rho.c and propagation constant k of a material from flow resistance information.formation.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.15 no.6 s.99
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• pp.680-686
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• 2005

This paper introduces the robust design of an intake system using transmission loss and the frequency weighting function. First, transmission loss is measured to evaluate the performance of the noise reduction for the intake system. The robust design parameters of the intake system are extracted by adapting a cost function with the Taguchi method. Subsequently, the frequency weighting function is developed by the subjective evaluation in which 6 special engineers were participated. Finally, the comparison between the proposed frequency weighted optimal design and unweighted optimal design for the transmission loss as the part is performed. Here, the overall levels of the transmission loss according to the methods are presented to validate the effectiveness of the proposed methodology.

• Transactions of the Korean Society of Automotive Engineers
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• v.6 no.4
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• pp.151-159
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• 1998

The heavy equipment such as tracter has been studied to improve rather the performance of engine than comfort. The mufflers of tracters have various specifications according to their uses. The exact analysis of various mufflers is needed to reduce the level of exhaust moise, a major noise source of engine, to improve the ride quality of tracter. In this study, a software based on Green's function is developed to predict the performance of sound transmission loss for a muffler according to the locations of inlet/outlet pipes. The locations of inlet and outlet pipes can be fixed at different position individually. The conventional muffler has the locations of inlet/outlet pipes on the direction of longitudinal axes. On other hand, the inlet and outlet pipes may be located at the circumferential surface of a test muffler such as one of tracter. The software is verified by analysis and experiment on current muffler of tracter and the improvement technique is proposed to reduce the level of exhaust noise.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.351-359
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• 2003

In this paper, we proposed a method for acoustic echo cancellation based on independent component analysis. When the large acoustic noise is picked up by the microphone, the performance of echo cancellation decreased. We used two microphones that received echo signal which is linearly mixed with the noise, then separated the echo signals from the received signals with independent component analysis algorithm. The separated echo signal is used for the reference signal of adaptive algorithm which leads to better performance of the echo cancellation. Computer simulation results show the validity of the proposed method.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.14 no.12
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• pp.1338-1346
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• 2004

The positions of rotating sound sources have been localized by experiments with the Doppler effects removed. In order to de-Dopplerize the sound signals emitted from moving sources, two kinds of signal reconstruction methods were applied. One is the forward propagation method and the other is the backward propagation method. Forward propagation method analyze the source emission time based on the instantaneous distance between sensors and the assumed source position, then the signals are reconstructed with respect to the emission time. On the other hand, the backward method uses time delay to do-Dopplerize the acquired data for the received time of reference. In both techniques. the reconstructed signal data were processed using beamforming algorithm to produce power distributions at the frequencies of interest. Experiments have been carried out for varying frequencies, rotating speeds and the object distances. It is shown that the forward propagation method gives better performance in locating source position than the backward propagation method.